Move ownership of RTPSenderAudio to ChannelSend.
This change takes out responsibility for packetization from the
RtpRtcp class, and deletes the method RtpRtcp::SendOutgoingData.
Video packetization was similarly moved in cl
https://webrtc-review.googlesource.com/c/src/+/123187
Bug: webrtc:7135
Change-Id: I0953125a5ca22a2ce51761b83693e0bb8ea74cd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27000}diff --git a/modules/rtp_rtcp/include/rtp_rtcp.h b/modules/rtp_rtcp/include/rtp_rtcp.h
index 8e1f98b..34ccfeb 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -146,11 +146,6 @@
// FEC/ULP/RED overhead (when FEC is enabled).
virtual size_t MaxRtpPacketSize() const = 0;
- virtual void RegisterAudioSendPayload(int payload_type,
- absl::string_view payload_name,
- int frequency,
- int channels,
- int rate) = 0;
virtual void RegisterSendPayloadFrequency(int payload_type,
int payload_frequency) = 0;
@@ -257,27 +252,6 @@
virtual RTPSender* RtpSender() = 0;
virtual const RTPSender* RtpSender() const = 0;
- // Used by the codec module to deliver a video or audio frame for
- // packetization.
- // |frame_type| - type of frame to send
- // |payload_type| - payload type of frame to send
- // |timestamp| - timestamp of frame to send
- // |payload_data| - payload buffer of frame to send
- // |payload_size| - size of payload buffer to send
- // |fragmentation| - fragmentation offset data for fragmented frames such
- // as layers or RED
- // |transport_frame_id_out| - set to RTP timestamp.
- // Returns true on success.
- virtual bool SendOutgoingData(FrameType frame_type,
- int8_t payload_type,
- uint32_t timestamp,
- int64_t capture_time_ms,
- const uint8_t* payload_data,
- size_t payload_size,
- const RTPFragmentationHeader* fragmentation,
- const RTPVideoHeader* rtp_video_header,
- uint32_t* transport_frame_id_out) = 0;
-
// Record that a frame is about to be sent. Returns true on success, and false
// if the module isn't ready to send.
virtual bool OnSendingRtpFrame(uint32_t timestamp,
@@ -432,23 +406,6 @@
const VideoBitrateAllocation& bitrate) = 0;
// **************************************************************************
- // Audio
- // **************************************************************************
-
- // Sends a TelephoneEvent tone using RFC 2833 (4733).
- // Returns -1 on failure else 0.
- virtual int32_t SendTelephoneEventOutband(uint8_t key,
- uint16_t time_ms,
- uint8_t level) = 0;
-
- // Store the audio level in dBov for header-extension-for-audio-level-
- // indication.
- // This API shall be called before transmision of an RTP packet to ensure
- // that the |level| part of the extended RTP header is updated.
- // return -1 on failure else 0.
- virtual int32_t SetAudioLevel(uint8_t level_dbov) = 0;
-
- // **************************************************************************
// Video
// **************************************************************************
diff --git a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 7d4f0f1..668d527 100644
--- a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -37,12 +37,6 @@
MOCK_METHOD1(SetRemoteSSRC, void(uint32_t ssrc));
MOCK_METHOD1(SetMaxRtpPacketSize, void(size_t size));
MOCK_CONST_METHOD0(MaxRtpPacketSize, size_t());
- MOCK_METHOD5(RegisterAudioSendPayload,
- void(int payload_type,
- absl::string_view payload_name,
- int frequency,
- int channels,
- int rate));
MOCK_METHOD2(RegisterSendPayloadFrequency,
void(int payload_type, int frequency));
MOCK_METHOD1(DeRegisterSendPayload, int32_t(int8_t payload_type));
@@ -87,16 +81,6 @@
uint32_t* nack_rate));
MOCK_CONST_METHOD1(EstimatedReceiveBandwidth,
int(uint32_t* available_bandwidth));
- MOCK_METHOD9(SendOutgoingData,
- bool(FrameType frame_type,
- int8_t payload_type,
- uint32_t timestamp,
- int64_t capture_time_ms,
- const uint8_t* payload_data,
- size_t payload_size,
- const RTPFragmentationHeader* fragmentation,
- const RTPVideoHeader* rtp_video_header,
- uint32_t* frame_id_out));
MOCK_METHOD4(OnSendingRtpFrame, bool(uint32_t, int64_t, int, bool));
MOCK_METHOD5(TimeToSendPacket,
bool(uint32_t ssrc,
@@ -162,9 +146,6 @@
MOCK_METHOD1(RegisterRtcpStatisticsCallback, void(RtcpStatisticsCallback*));
MOCK_METHOD0(GetRtcpStatisticsCallback, RtcpStatisticsCallback*());
MOCK_METHOD1(SendFeedbackPacket, bool(const rtcp::TransportFeedback& packet));
- MOCK_METHOD3(SendTelephoneEventOutband,
- int32_t(uint8_t key, uint16_t time_ms, uint8_t level));
- MOCK_METHOD1(SetAudioLevel, int32_t(uint8_t level_dbov));
MOCK_METHOD1(SetTargetSendBitrate, void(uint32_t bitrate_bps));
MOCK_METHOD1(SetKeyFrameRequestMethod, int32_t(KeyFrameRequestMethod method));
MOCK_METHOD0(RequestKeyFrame, int32_t());
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 4e06002..6d1c9e2 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -116,9 +116,7 @@
configuration.extmap_allow_mixed,
configuration.field_trials ? *configuration.field_trials
: default_trials));
- if (configuration.audio) {
- audio_ = absl::make_unique<RTPSenderAudio>(clock_, rtp_sender_.get());
- }
+
// Make sure rtcp sender use same timestamp offset as rtp sender.
rtcp_sender_.SetTimestampOffset(rtp_sender_->TimestampOffset());
@@ -270,17 +268,6 @@
rtcp_receiver_.IncomingPacket(rtcp_packet, length);
}
-void ModuleRtpRtcpImpl::RegisterAudioSendPayload(int payload_type,
- absl::string_view payload_name,
- int frequency,
- int channels,
- int rate) {
- RTC_DCHECK(audio_);
- rtcp_sender_.SetRtpClockRate(payload_type, frequency);
- RTC_CHECK_EQ(0, audio_->RegisterAudioPayload(payload_name, payload_type,
- frequency, channels, rate));
-}
-
void ModuleRtpRtcpImpl::RegisterSendPayloadFrequency(int payload_type,
int payload_frequency) {
rtcp_sender_.SetRtpClockRate(payload_type, payload_frequency);
@@ -425,30 +412,6 @@
rtp_sender_->SetAsPartOfAllocation(part_of_allocation);
}
-bool ModuleRtpRtcpImpl::SendOutgoingData(
- FrameType frame_type,
- int8_t payload_type,
- uint32_t time_stamp,
- int64_t capture_time_ms,
- const uint8_t* payload_data,
- size_t payload_size,
- const RTPFragmentationHeader* fragmentation,
- const RTPVideoHeader* rtp_video_header,
- uint32_t* transport_frame_id_out) {
- OnSendingRtpFrame(time_stamp, capture_time_ms, payload_type,
- kVideoFrameKey == frame_type);
-
- const uint32_t rtp_timestamp = time_stamp + rtp_sender_->TimestampOffset();
- if (transport_frame_id_out)
- *transport_frame_id_out = rtp_timestamp;
-
- RTC_DCHECK(audio_);
- RTC_DCHECK(fragmentation == nullptr);
-
- return audio_->SendAudio(frame_type, payload_type, rtp_timestamp,
- payload_data, payload_size);
-}
-
bool ModuleRtpRtcpImpl::OnSendingRtpFrame(uint32_t timestamp,
int64_t capture_time_ms,
int payload_type,
@@ -787,17 +750,6 @@
return rtcp_sender_.SendFeedbackPacket(packet);
}
-// Send a TelephoneEvent tone using RFC 2833 (4733).
-int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(const uint8_t key,
- const uint16_t time_ms,
- const uint8_t level) {
- return audio_ ? audio_->SendTelephoneEvent(key, time_ms, level) : -1;
-}
-
-int32_t ModuleRtpRtcpImpl::SetAudioLevel(const uint8_t level_d_bov) {
- return audio_ ? audio_->SetAudioLevel(level_d_bov) : -1;
-}
-
int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod(
const KeyFrameRequestMethod method) {
key_frame_req_method_ = method;
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 20eb179..4b1c927 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -32,8 +32,6 @@
#include "modules/rtp_rtcp/source/rtcp_receiver.h"
#include "modules/rtp_rtcp/source/rtcp_sender.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
-#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
-#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/gtest_prod_util.h"
@@ -64,12 +62,6 @@
void SetRemoteSSRC(uint32_t ssrc) override;
// Sender part.
-
- void RegisterAudioSendPayload(int payload_type,
- absl::string_view payload_name,
- int frequency,
- int channels,
- int rate) override;
void RegisterSendPayloadFrequency(int payload_type,
int payload_frequency) override;
@@ -137,18 +129,6 @@
void SetAsPartOfAllocation(bool part_of_allocation) override;
- // Used by the codec module to deliver a video or audio frame for
- // packetization.
- bool SendOutgoingData(FrameType frame_type,
- int8_t payload_type,
- uint32_t time_stamp,
- int64_t capture_time_ms,
- const uint8_t* payload_data,
- size_t payload_size,
- const RTPFragmentationHeader* fragmentation,
- const RTPVideoHeader* rtp_video_header,
- uint32_t* transport_frame_id_out) override;
-
bool OnSendingRtpFrame(uint32_t timestamp,
int64_t capture_time_ms,
int payload_type,
@@ -270,17 +250,6 @@
bool RtcpXrRrtrStatus() const override;
- // Audio part.
-
- // Send a TelephoneEvent tone using RFC 2833 (4733).
- int32_t SendTelephoneEventOutband(uint8_t key,
- uint16_t time_ms,
- uint8_t level) override;
-
- // Store the audio level in d_bov for header-extension-for-audio-level-
- // indication.
- int32_t SetAudioLevel(uint8_t level_d_bov) override;
-
// Video part.
// Set method for requesting a new key frame.
@@ -346,7 +315,6 @@
bool TimeToSendFullNackList(int64_t now) const;
std::unique_ptr<RTPSender> rtp_sender_;
- std::unique_ptr<RTPSenderAudio> audio_;
RTCPSender rtcp_sender_;
RTCPReceiver rtcp_receiver_;
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index da541a5..98067b2 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -22,6 +22,7 @@
#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl.h"
+#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "rtc_base/rate_limiter.h"
#include "test/gmock.h"
#include "test/gtest.h"
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.cc b/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 56d0884..c049530 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -47,7 +47,9 @@
} // namespace
RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender)
- : clock_(clock), rtp_sender_(rtp_sender) {}
+ : clock_(clock), rtp_sender_(rtp_sender) {
+ RTC_DCHECK(clock_);
+}
RTPSenderAudio::~RTPSenderAudio() {}